U.S. patent application number 12/750175 was filed with the patent office on 2010-12-02 for enhanced group calling features for connected portfolio services in a wireless communications network.
This patent application is currently assigned to KODIAK NETWORKS, INC.. Invention is credited to Basem A. Ardah, Ravi Ayyasamy, Shiva K. K. Cheedella, Ramu Kandula, Ravi Shankar Kumar, Gorachand Kundu, Bruce D. Lawler, Harisha M. Negalaguli, Chetan M. Patel, Krishnakant M. Patel, Brahmananda R. Vempati.
Application Number | 20100304724 12/750175 |
Document ID | / |
Family ID | 42936503 |
Filed Date | 2010-12-02 |
United States Patent
Application |
20100304724 |
Kind Code |
A1 |
Lawler; Bruce D. ; et
al. |
December 2, 2010 |
ENHANCED GROUP CALLING FEATURES FOR CONNECTED PORTFOLIO SERVICES IN
A WIRELESS COMMUNICATIONS NETWORK
Abstract
Enhanced Group Calling Features for Connected Portfolio Services
in wireless communications networks, such as a mobile or cellular
phone communications networks. The Connected Portfolio Services
include Mobile Conferencing (Scheduled/Instant/Reservationless
Conference), Family Connect, Buddy Connect, and Quick Reach, while
the Enhanced Group Calling Features include Voicemail Diversion
(the recognition of a diversion to voicemail and dropping the leg),
Reverse Quick Reach (where a terminating subscriber defines how a
call dialed to his/her mobile number should be handled and
directed), and Single Number Group Calling (an optimal design
implementation of single number based group calling).
Inventors: |
Lawler; Bruce D.; (Kirkwood,
CA) ; Patel; Krishnakant M.; (Richardson, TX)
; Ayyasamy; Ravi; (Richardson, TX) ; Negalaguli;
Harisha M.; (Plano, TX) ; Ardah; Basem A.;
(Plano, TX) ; Kundu; Gorachand; (Bangalore,
IN) ; Kandula; Ramu; (Bangalore, IN) ;
Vempati; Brahmananda R.; (Dallas, TX) ; Kumar; Ravi
Shankar; (Plano, TX) ; Patel; Chetan M.;
(Richardson, TX) ; Cheedella; Shiva K. K.;
(Houston, TX) |
Correspondence
Address: |
GATES & COOPER LLP;HOWARD HUGHES CENTER
6701 CENTER DRIVE WEST, SUITE 1050
LOS ANGELES
CA
90045
US
|
Assignee: |
KODIAK NETWORKS, INC.
San Ramon
CA
|
Family ID: |
42936503 |
Appl. No.: |
12/750175 |
Filed: |
March 30, 2010 |
Related U.S. Patent Documents
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Application
Number |
Filing Date |
Patent Number |
|
|
61164754 |
Mar 30, 2009 |
|
|
|
61172129 |
Apr 23, 2009 |
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Current U.S.
Class: |
455/414.1 |
Current CPC
Class: |
H04M 3/42229 20130101;
H04M 3/46 20130101; H04M 2207/18 20130101; H04M 3/42 20130101; H04L
65/403 20130101; H04W 4/06 20130101; H04M 2203/2033 20130101 |
Class at
Publication: |
455/414.1 |
International
Class: |
H04W 4/00 20090101
H04W004/00 |
Claims
1. An apparatus for providing connected portfolio services in a
wireless network, comprising: a wireless network for making calls
between handsets, wherein the calls are initiated by call setup and
in-band signaling within the wireless network and voice frames for
the calls are switched between the handsets by at least one mobile
switching center across bearer paths in the wireless network; and a
real-time exchange that interfaces to at least one mobile switching
center in the wireless network to provide connected portfolio
services therein, without requiring any changes to the mobile
switching center or other equipment of the wireless network to
provide the connected portfolio services; wherein both the
real-time exchange and the handsets that use the connected
portfolio services communicate with each other using the call setup
and in-band signaling within the wireless network, such that at
least one mobile switching center routes an originating leg of the
connected portfolio services from an originating handset to the
real-time exchange, the real-time exchange initiates one or more
terminating legs of the connected portfolio services to one or more
terminating handsets through at least one mobile switching center,
and the real-time exchange switches the voice frames for the
connected portfolio services from the originating handset to the
terminating handsets across the bearer paths and through at least
one mobile switching center that switches the voice frames for both
the calls and the connected portfolio services in the wireless
network; and wherein the real-time exchange drops one or more of
the terminating legs that has been diverted to voice mail from the
connected portfolio services.
2. The apparatus of claim 1, wherein the real-time exchange
recognizes that the one or more of the terminating legs has been
diverted to voice mail when a diversion indicator is received by
the real-time exchange.
3. The apparatus of claim 2, wherein the diversion indicator
includes a diversion number that is compared by the real-time
exchange to a list of voice mail numbers.
4. The apparatus of claim 1, wherein the real-time exchange
recognizes that the one or more of the terminating legs of the
advanced voice services has been diverted to voice mail when no
indicator confirming participation is received by the real-time
exchange.
5. The apparatus of claim 4, wherein the indicator confirming
participation comprises a key on the handset pressed by a user upon
answering the terminating leg to confirm participation in the
connected portfolio services.
6. An apparatus for providing connected portfolio services in a
wireless network, comprising: a wireless network for making calls
between handsets, wherein the calls are initiated by call setup and
in-band signaling within the wireless network and voice frames for
the calls are switched between the handsets by at least one mobile
switching center across bearer paths in the wireless network; and a
real-time exchange that interfaces to at least one mobile switching
center in the wireless network to provide connected portfolio
services therein, without requiring any changes to the mobile
switching center or other equipment of the wireless network to
provide the connected portfolio services; wherein both the
real-time exchange and the handsets that use the connected
portfolio services communicate with each other using the call setup
and in-band signaling within the wireless network, such that at
least one mobile switching center routes an originating leg of the
connected portfolio services from an originating handset to the
real-time exchange, the real-time exchange initiates one or more
terminating legs of the connected portfolio services to one or more
terminating handsets through at least one mobile switching center,
and the real-time exchange switches the voice frames for the
connected portfolio services from the originating handset to the
terminating handsets across the bearer paths and through at least
one mobile switching center that switches the voice frames for both
the calls and the connected portfolio services in the wireless
network; and wherein the real-time exchange initiates a plurality
of the terminating legs to a plurality of numbers configured for
one of the terminating handsets.
7. The apparatus of claim 6, wherein a subscriber for the
terminating handset defines the plurality of numbers to be dialed
when the advanced voice services is directed to the terminating
handset.
8. The apparatus of claim 6, wherein the real-time exchange
initiates the plurality of terminating legs to the plurality of
numbers simultaneously or sequentially.
9. The apparatus of claim 6, wherein, when one of the plurality of
terminating legs is answered, the real-time exchange drops
unanswered ones of the plurality of terminating legs.
10. The apparatus of claim 6, wherein, when one of the plurality of
terminating legs is answered, and an indicator confirming
participation is received by the real-time exchange, the real-time
exchange drops unanswered ones of the plurality of terminating
legs.
11. The apparatus of claim 10, wherein the indicator confirming
participation comprises a key on the handset pressed by a user upon
answering the terminating leg to confirm participation in the
connected portfolio services.
12. The apparatus of claim 6, wherein, when one of the plurality of
terminating legs is answered, the real-time exchange excludes
itself from the bearer path, when there is a single terminating leg
connected to the originating leg.
13. An apparatus for providing connected portfolio services in a
wireless network, comprising: a wireless network for making calls
between handsets, wherein the calls are initiated by call setup and
in-band signaling within the wireless network and voice frames for
the calls are switched between the handsets by at least one mobile
switching center across bearer paths in the wireless network; and a
real-time exchange that interfaces to at least one mobile switching
center in the wireless network to provide connected portfolio
services therein, without requiring any changes to the mobile
switching center or other equipment of the wireless network to
provide the connected portfolio services; wherein both the
real-time exchange and the handsets that use the connected
portfolio services communicate with each other using the call setup
and in-band signaling within the wireless network, such that at
least one mobile switching center routes an originating leg of the
connected portfolio services from an originating handset to the
real-time exchange, the real-time exchange initiates one or more
terminating legs of the connected portfolio services to one or more
terminating handsets through at least one mobile switching center,
and the real-time exchange switches the voice frames for the
connected portfolio services from the originating handset to the
terminating handsets across the bearer paths and through at least
one mobile switching center that switches the voice frames for both
the calls and the connected portfolio services in the wireless
network; and wherein the real-time exchange allocates a single
number to represent a logical group of numbers for the terminating
legs of the connected portfolio services.
14. The apparatus of claim 13, wherein a call addressed to the
single number is distributed to one of a plurality of real-time
exchanges deployed in the network.
15. The apparatus of claim 14, wherein the single number is
translated to a temporary routing number to re-direct the call to
the real-time exchange designated for handling the single number or
re-direct the call to a geographically redundant real-time
exchange.
16. The apparatus of claim 15, wherein the temporary routing number
is used by the real-time exchange to retrieve an original number
and the logical group of numbers.
17. The apparatus of claim 13, further comprising a redundant
real-time exchange positioned at a geographically different
location that performs the same functions as the real-time exchange
when the real-time exchange is unavailable.
Description
CROSS REFERENCE TO RELATED APPLICATIONS
[0001] This application claims the benefit under 35 U.S.C. Section
119(e) of the following co-pending and commonly-assigned patent
application:
[0002] U.S. Provisional Application Ser. No. 61/164,754, filed Mar.
30, 2009, by Bruce D. Lawler, Krishnakant M. Patel, Ravi Ayyasamy,
Harisha Mahabaleshwara Negalaguli, Basem A. Ardah, Gorachand Kundu,
Ramu Kandula, and Brahmananda R. Vempati, entitled "ENHANCED GROUP
CALLING FEATURES," attorneys' docket number 154.39-US-P1, and
[0003] U.S. Provisional Application Ser. No. 61/172,129, filed Apr.
23, 2009, by Krishnakant M. Patel, Ravi Ayyasamy, Harisha
Mahabaleshwara Negalaguli, Ravi Shankar Kumar, and Chetan Patel,
entitled "IP CONNECTIVITY FOR CONNECTED PORTFOLIO APPLICATIONS,"
attorneys' docket number 154.39-US-P2,
[0004] which applications are incorporated by reference herein.
[0005] This application is related to the following co-pending and
commonly-assigned patent applications:
[0006] U.S. application Ser. No. 10/515,556, filed Nov. 23, 2004,
by Gorachand Kundu, Ravi Ayyasamy and Krishnakant Patel, entitled
"DISPATCH SERVICE ARCHITECTURE FRAMEWORK," attorney docket number
G&C 154.4-US-WO, which application claims the benefit under 35
U.S.C. Section 365 of P.C.T. International Application Serial
Number PCT/US03/16386 (154.4-WO-U1), which application claims the
benefit under 35 U.S.C. Section 119(e) of U.S. Provisional
Application Ser. Nos. 60/382,981 (154.3-US-P1), 60/383,179
(154.4-US-P1) and 60/407,168 (154.5-US-P1);
[0007] U.S. application Ser. No. 10/564,903, filed Jan. 17, 2006,
by F. Craig Farrill, Bruce D. Lawler and Krishnakant M. Patel,
entitled "PREMIUM VOICE SERVICES FOR WIRELESS COMMUNICATIONS
SYSTEMS," attorney docket number G&C 154.7-US-WO, which
application claims the benefit under 35 U.S.C. Section 365 of
P.C.T. International Application Serial Number PCT/US04/23038
(154.7-WO-U1), which application claims the benefit under 35 U.S.C.
Section 119(e) of U.S. Provisional Application Ser. Nos. 60/488,638
(154.7-US-P1), 60/492,650 (154.8-US-P1) and 60/576,094
(154.14-US-P1) and which application is a continuation-in-part and
claims the benefit under 35 U.S.C. Sections 119, 120 and/or 365 of
P.C.T. International Application Serial Number PCT/US03/16386
(154.4-WO-U1);
[0008] U.S. patent application Ser. No. 11/126,587, filed May 11,
2005, by Ravi Ayyasamy and Krishnakant M. Patel, entitled
"ARCHITECTURE, CLIENT SPECIFICATION AND APPLICATION PROGRAMMING
INTERFACE (API) FOR SUPPORTING ADVANCED VOICE SERVICES (AVS)
INCLUDING PUSH TO TALK ON WIRELESS HANDSETS AND NETWORKS," attorney
docket number 154.9-US-U1, which application claims the benefit
under 35 U.S.C. Section 119(e) of U.S. Provisional Application Ser.
Nos. 60/569,953 (154.9-US-P1) and 60/579,309 (154.15-US-P1), and
which application is a continuation-in-part and claims the benefit
under 35 U.S.C. Sections 119, 120 and/or 365 of U.S. application
Ser. No. 10/515,556 (154.4-US-WO) and P.C.T. International
Application Serial Number PCT/US04/23038 (154.7-WO-U1);
[0009] U.S. application Ser. No. 11/129,268, filed May 13, 2005, by
Krishnakant M. Patel, Gorachand Kundu, Ravi Ayyasamy and Basem
Ardah, entitled "ROAMING GATEWAY FOR SUPPORT OF ADVANCED VOICE
SERVICES WHILE ROAMING IN WIRELESS COMMUNICATIONS SYSTEMS,"
attorney docket number 154.10-US-U1, now U.S. Pat. No. 7,403,775,
issued Jul. 22, 2008, which application claims the benefit under 35
U.S.C. Section 119(e) of U.S. Provisional Application Ser. No.
60/571,075 (154.10-US-P1), and which application is a
continuation-in-part and claims the benefit under 35 U.S.C.
Sections 119, 120 and/or 365 of U.S. application Ser. No.
10/515,556 (154.4-US-WO) and P.C.T. International Application
Serial Number PCT/US04/23038 (154.7-WO-U1);
[0010] U.S. application Ser. No. 11/134,883, filed May 23, 2005, by
Krishnakant Patel, Vyankatesh V. Shanbhag, Ravi Ayyasamy, Stephen
R. Horton and Shan-Jen Chiou, entitled "ADVANCED VOICE SERVICES
ARCHITECTURE FRAMEWORK," attorney docket number 154.11-US-U1, which
application claims the benefit under 35 U.S.C. Section 119(e) of
U.S. Provisional Application Ser. Nos. 60/573,059 (154.11-US-P1)
and 60/576,092 (154.12-US-P1), and which application is a
continuation-in-part and claims the benefit under 35 U.S.C.
Sections 119, 120 and/or 365 of U.S. application Ser. No.
10/515,556 (154.4-US-WO), P.C.T. International Application Serial
Number PCT/US04/23038 (154.7-WO-U1), U.S. application Ser. No.
11/126,587 (154.9-US-U1), and U.S. application Ser. No. 11/129,268
(154.10-US-U1);
[0011] U.S. application Ser. No. 11/136,233, filed May 24, 2005, by
Krishnakant M. Patel, Vyankatesh Vasant Shanbhag, and Anand
Narayanan, entitled "SUBSCRIBER IDENTITY MODULE (SIM) ENABLING
ADVANCED VOICE SERVICES (AVS) INCLUDING PUSH-TO-TALK,
PUSH-TO-CONFERENCE AND PUSH-TO-MESSAGE ON WIRELESS HANDSETS AND
NETWORKS," attorney docket number 154.13-US-U1, which application
claims the benefit under 35 U.S.C. Section 119(e) of U.S.
Provisional Application Ser. No. 60/573,780 (154.13-US-P1), and
which application is a continuation-in-part and claims the benefit
under 35 U.S.C. Sections 119, 120 and/or 365 of U.S. application
Ser. No. 10/515,556 (154.4-US-WO), P.C.T. International Application
Serial Number PCT/US04/23038 (154.7-WO-U1), U.S. application Ser.
No. 11/126,587 (154.9-US-U1), and U.S. application Ser. No.
11/134,883 (154.11-US-U1);
[0012] U.S. application Ser. No. 11/158,527, filed Jun. 22, 2005,
by F. Craig Farrill, entitled "PRESS-TO-CONNECT FOR WIRELESS
COMMUNICATIONS SYSTEMS," attorney docket number 154.16-US-U1, which
application claims the benefit under 35 U.S.C. Section 119(e) of
U.S. Provisional Application Ser. No. 60/581,954 (154.16-US-P1),
and which application is a continuation-in-part and claims the
benefit under 35 U.S.C. Sections 119, 120 and/or 365 of U.S.
application Ser. No. 10/515,556 (154.4-US-WO) and P.C.T.
International Application Serial Number PCT/US4/23038
(154.7-WO-U1);
[0013] U.S. application Ser. No. 11/183,516, filed Jul. 18, 2005,
by Deepankar Biswaas, entitled "VIRTUAL PUSH TO TALK (PTT) AND PUSH
TO SHARE (PTS) FOR WIRELESS COMMUNICATIONS SYSTEMS," attorney
docket number 154.17-US-U1, which application claims the benefit
under 35 U.S.C. Section 119(e) of U.S. Provisional Application Ser.
No. 60/588,464 (154.17-US-P1);
[0014] U.S. application Ser. No. 11/356,775, filed Feb. 17, 2006,
by Krishnakant M. Patel, Bruce D. Lawler, Giridhar K. Boray, and
Brahmananda R. Vempati, entitled "ENHANCED FEATURES IN AN ADVANCED
VOICE SERVICES (AVS) FRAMEWORK FOR WIRELESS COMMUNICATIONS
SYSTEMS," attorney docket number 154.18-US-U1, which application
claims the benefit under 35 U.S.C. Section 119(e) of U.S.
Provisional Application Ser. No. 60/654,271(154.18-US-P1); P.C.T.
International Application Serial Number PCT/US2006/011628, filed
Mar. 30, 2006, by Krishnakant M. Patel, Gorachand Kundu, Sameer
Dharangaonkar, Giridhar K. Boray, and Deepankar Biswas, entitled
"TECHNIQUE FOR IMPLEMENTING ADVANCED VOICE SERVICES USING AN
UNSTRUCTURED SUPPLEMENTARY SERVICE DATA (USSD) INTERFACE," attorney
docket number 154.19-WO-U1, which application claims the benefit
under 35 U.S.C. Section 119(e) of U.S. Provisional Application Ser.
No. 60/666,424 (154.19-US-P1);
[0015] U.S. application Ser. No. 11/462,332, filed Aug. 3, 2006, by
Deepankar Biswas, Krishnakant M. Patel, Giridhar K. Boray, and
Gorachand Kundu, entitled "ARCHITECTURE AND IMPLEMENTATION OF
CLOSED USER GROUP AND LIMITING MOBILITY IN WIRELESS NETWORKS,"
attorney docket number 154.20-US-U1, which application claims the
benefit under 35 U.S.C. Section 119(e) of U.S. Provisional
Application Ser. No. 60/705,115 (154.20-US-P1);
[0016] U.S. application Ser. No. 11/463,186, filed Aug. 8, 2006, by
Ravi Ayyasamy and Krishnakant M. Patel, entitled "ADVANCED VOICE
SERVICES CLIENT FOR BREW PLATFORM," attorney docket number
154.21-US-U1, which application claims the benefit under 35 U.S.C.
Section 119(e) of U.S. Provisional Application Ser. No. 60/706,265
(154.21-US-P1);
[0017] U.S. application Ser. No. 11/567,098, filed Dec. 5, 2006, by
Ravi Ayyasamy, Bruce D. Lawler, Krishnakant M. Patel, Vyankatesh V.
Shanbhag, Brahmananda R. Vempati, and Ravi Shankar Kumar, entitled
"INSTANT MESSAGING INTERWORKING IN AN ADVANCED VOICE SERVICES (AVS)
FRAMEWORK FOR WIRELESS COMMUNICATIONS SYSTEMS," attorney docket
number 154.23-US-U1, which application claims the benefit under 35
U.S.C. Section 119(e) of U.S. Provisional Application Ser. No.
60/742,250 (154.23-US-P1);
[0018] U.S. application Ser. No. 11/740,805, filed Apr. 26, 2007,
by Krishnakant M. Patel, Giridhar K. Boray, Ravi Ayyasamy, and
Gorachand Kundu, entitled "ADVANCED FEATURES ON A REAL-TIME
EXCHANGE SYSTEM," attorney docket number 154.26-US-U1, which
application claims the benefit under 35 U.S.C. Section 119(e) of
U.S. Provisional Application Ser. No. 60/795,090
(154.26-US-P1);
[0019] U.S. application Ser. No. 11/891,127, filed Aug. 9, 2007, by
Krishnakant M. Patel, Deepankar Biswas, Sameer P. Dharangaonkar and
Terakanambi Nanjanayaka Raja, entitled "EMERGENCY GROUP CALLING
ACROSS MULTIPLE WIRELESS NETWORKS," attorney docket number
154.27-US-U1, which application claims the benefit under 35 U.S.C.
Section 119(e) of U.S. Provisional Application Ser. No. 60/836,521
(154.27-US-P1);
[0020] U.S. application Ser. No. 12/259,102, filed Oct. 27, 2008,
by Krishnakant M. Patel, Bruce Lawler, Gorachand Kundu, Ravi
Ayyasamy, Ravi Shankar Kumar, Harisha Mahabaleshwara Negalaguli,
Basem Ahmad Ardah, Prathap Chandana, Shan-Jen Chiou, Arun
Velayudhan, and Ramu Kandula, entitled "CONNECTED PORTFOLIO
SERVICES FOR A WIRELESS COMMUNICATIONS NETWORK," attorneys' docket
number 154.32-US-U1, which application claims the benefit under 35
U.S.C. Section 119(e) of U.S. Provisional Application Ser. No.
60/982,650 (154.32-US-P1) and U.S. Provisional Application Ser. No.
61/023,042 (154.32-US-P2);
[0021] U.S. application Ser. No. 12/359,861, filed Jan. 26, 2009,
by Bruce D. Lawler, Krishnakant M. Patel, Ravi Ayyasamy, Harisha
Mahabaleshwara Negalaguli, Binu Kaiparambil, Shiva Cheedella,
Brahmananda R. Vempati, Ravi Shankar Kumar, and Avrind Shanbhag,
entitled "CONVERGED MOBILE-WEB COMMUNICATIONS SOLUTION," attorney
docket number 154.33-US-U1, which application claims the benefit
under 35 U.S.C. Section 119(e) of U.S. Provisional Application Ser.
No. 61/023,332 (154.33-US-P1);
[0022] U.S. application Ser. No. 12/582,601, filed Oct. 20, 2009,
by Krishnakant M. Patel, Ravi Ayyasamy, Gorachand Kundu, Basem A.
Ardah, Anand Narayanan, Brahmananda R. Vempati, and Pratap
Chandana, entitled "HYBRID PUSH-TO-TALK FOR MOBILE PHONE NETWORKS,"
attorney docket number 154.36-US-U1, which application claims the
benefit under 35 U.S.C. Section 119(e) of U.S. Provisional
Application Ser. No. 61/106,689 (154.36-US-P1);
[0023] all of which applications are incorporated by reference
herein.
BACKGROUND OF THE INVENTION
[0024] 1. Field of the Invention
[0025] This invention relates in general to wireless communications
systems, and more specifically, to enhanced group calling features
for connected portfolio services in a wireless communications
network.
[0026] 2. Description of Related Art
[0027] Advanced Voice Services (AVS), also known as Advanced Group
Services (AGS), can include a number of different functions, such
as Push-to-Conference (P2C) or Instant Conferencing, etc., as
described in the co-pending and commonly-assigned patent
applications cross-referenced above and incorporated by reference
herein. These AVS functions have enormous revenue earnings
potential for wireless communications systems, such as mobile phone
networks.
[0028] Currently, there are three major approaches employed in
providing AVS and AGS in wireless communications systems. One
approach requires the installation of a dedicated private network,
parallel to the wireless communications system, to support the
group-based voice services. However, a dedicated private network is
costly to install and maintain.
[0029] Another approach is based on Voice over IP (VoIP)
technologies. While this approach promises compliance with newer
and emerging standards, such as GPRS (General Packet Radio
Service), UMTS (Universal Mobile Telecommunications System), etc.,
it does not provide a solution for carriers employing wireless
communications systems based on existing standards, such as GSM,
CDMA, etc. However, even for the newer standards, solutions based
on VoIP have serious drawbacks, including slower call setup,
significant overhead, increased susceptibility to packet losses,
low bit rate voice coders, and significant modifications to the
mobile handset.
[0030] Still another approach is the innovative approach described
in the co-pending and commonly-assigned patent applications
cross-referenced above and incorporated by reference herein. In
this approach, advanced voice services are provided by a real-time
exchange (RTX), also known as a dispatch gateway (DG), that
interfaces to the wireless communications system to provide the
advanced voice services therein, wherein both the real-time
exchange and mobiles that use the advanced voice services
communicate with each other using call setup and in-band signaling
within the wireless communications system.
[0031] However, notwithstanding the innovations described in the
co-pending and commonly-assigned patent applications
cross-referenced above, there is a need in the art for improvements
to the AVS, as well as additional AVS, that comply with existing
and emerging wireless standards and provide superior user
experiences. The present invention aims to satisfy this need by
providing additional services, also known as Connected Portfolio
Services, and improvements to those additional services, also known
as Enhanced Group Calling Features for Connected Portfolio
Services, in wireless communications systems.
SUMMARY OF THE INVENTION
[0032] To overcome the limitations in the prior art described
above, and to overcome other limitations that will become apparent
upon reading and understanding the present specification, the
present invention discloses Enhanced Group Calling Features for
Connected Portfolio Services in wireless communications networks,
such as a mobile or cellular phone communications networks. The
Connected Portfolio Services include Mobile Conferencing
(Scheduled/Instant/Reservationless Conference), Family Connect,
Buddy Connect, and Quick Reach, while the Enhanced Group Calling
Features include Voicemail Diversion, Reverse Quick Reach, and
Single Number Group Calling. These and other aspects of the present
invention are described in more detail below.
BRIEF DESCRIPTION OF THE DRAWINGS
[0033] Referring now to the drawings in which like reference
numbers represent corresponding parts throughout:
[0034] FIG. 1 is a block diagram that illustrates an exemplary
embodiment of a wireless communications network, according to a
preferred embodiment of the present invention.
[0035] FIG. 2 illustrates a proposed architecture for the RTX,
according to the preferred embodiment of the present invention.
[0036] FIG. 3 illustrates the high-level functional components and
their interfaces in a handset, according to a preferred embodiment
of the present invention.
[0037] FIG. 4 illustrates the user interface for a conference
scheduler as displayed on the handset, according to a preferred
embodiment of the present invention.
[0038] FIG. 5 is a flowchart that illustrates the steps performed
in a Scheduled Conference, according to a preferred embodiment of
the present invention. FIG. 6 is a flowchart that illustrates the
steps performed in a Reservationless
[0039] Conference Origination, according to a preferred embodiment
of the present invention.
[0040] FIG. 7 illustrates a first method for dropping a voicemail
leg in a group calling scenario, according to a preferred
embodiment of the present invention.
[0041] FIG. 8 illustrates a second method for dropping a voicemail
leg in a group calling scenario, according to a preferred
embodiment of the present invention.
[0042] FIG. 9 illustrates a method for performing Reverse Quick
Reach, according to a preferred embodiment of the present
invention.
[0043] FIG. 10 illustrates a method for performing Quick Reach and
Reverse Quick Reach optimization, according to a preferred
embodiment of the present invention.
[0044] FIG. 11 illustrates a method for performing the single
number based group calling in a normal scenario when the RTX is
active, according to a preferred embodiment of the present
invention.
[0045] FIG. 12 illustrates a method for performing the single
number based group calling in a geo-redundancy approach when the
RTX is unavailable, according to a preferred embodiment of the
present invention.
[0046] FIG. 13 illustrates a method for performing the single
number based group calling in a geo-redundancy approach when the
RTX is unavailable and a "race condition" exists due to a second
call from the same user, according to a preferred embodiment of the
present invention.
[0047] FIG. 14 illustrates a method for performing the single
number based group calling in a normal scenario when the RTX is
active, according to a preferred embodiment of the present
invention.
[0048] FIG. 15 illustrates a method for performing the single
number based group calling in a normal scenario when the RTX is
active, according to a preferred embodiment of the present
invention.
[0049] FIG. 16 illustrates a method for performing the single
number based group calling in a geo-redundancy approach when the
RTX is unavailable, according to a preferred embodiment of the
present invention.
[0050] FIG. 17 illustrates a method for performing the single
number based group calling in a geo-redundancy approach when the
RTX is active, according to a preferred embodiment of the present
invention.
[0051] FIG. 18 illustrates a method for performing the single
number based group calling in a geo-redundancy approach when the
RTX is unavailable, according to a preferred embodiment of the
present invention.
DETAILED DESCRIPTION OF THE INVENTION
[0052] In the following description of the preferred embodiment,
reference is made to the accompanying drawings which form a part
hereof, and in which is shown by way of illustration the specific
embodiment in which the invention may be practiced. It is to be
understood that other embodiments may be utilized as structural
changes may be made without departing from the scope of the present
invention.
[0053] 1 Overview
[0054] 1.1 Enhanced Group Calling Features for Connected Portfolio
Services in a Wireless Communications Network
[0055] The present invention discloses Connected Portfolio
Services, which are also described in co-pending and commonly
assigned U.S. application Ser. No. 12/259,102, filed Oct. 27, 2008,
by Krishnakant M. Patel, Bruce Lawler, Gorachand Kundu, Ravi
Ayyasamy, Ravi Shankar Kumar, Harisha Mahabaleshwara Negalaguli,
Basem Ahmad Ardah, Prathap Chandana, Shan-Jen Chiou, Arun
Velayudhan, and Ramu Kandula, entitled "CONNECTED PORTFOLIO
SERVICES FOR A WIRELESS COMMUNICATIONS NETWORK," attorneys' docket
number 154.32-US-U1, which application claims the benefit under 35
U.S.C. Section 119(e) of U.S. Provisional Application Ser. No.
60/982,650 (154.32-US-P1) and U.S. Provisional Application Ser. No.
61/023,042 (154.32-US-P2), which applications are incorporated by
reference herein. The present invention also discloses improvement
to the Connected Portfolio Services, known as Enhanced Group
Calling Features for Connected Portfolio Services in wireless
communications systems, which are described in more detail
herein.
[0056] More specifically, the Connected Portfolio Services include
Mobile Conferencing (Scheduled/Instant/Reservationless Conference),
Family Connect, Buddy Connect, and Quick Reach:
[0057] 1. Scheduled Conference: This service allows a mobile
handset user to schedule a conference with a group of other users
at a predetermined date and time. There are two modes of operation:
Dial-Out and Dial-In:
[0058] a) Dial-Out: In this option, a Real-Time Exchange (RTX) will
dial out the call to participants in a scheduled conference, and
then bridge the conference call between the participants.
[0059] b) Dial-In: In this option, participants in a scheduled
conference dial in to a conference bridge number.
[0060] A number of unique technologies are provided to facilitate
the Scheduled Conference service with many user friendly features,
such as conference call notification, one-touch dial to join a
conference call, etc.
[0061] 2. Reservationless Conference: This service allows a user to
set up a conference bridge and communicate the conference bridge
access number and password to participants of a conference call.
This solution is "clientless" in that the originator does not need
a handset client application to invoke this service.
[0062] 3. Instant Conferencing: This service allows users to create
and manage groups using multiple different means, such as via the
Web, via Short Message Service (SMS), via Wireless Access Protocol
(WAP), via an operator, etc. Once a group is created, the
originator receives a single dial out number; upon dialing this
number, the originator is connected to the group.
[0063] 4. Family Connect: The Family Connect service allows user to
make an instant conference call to all the family members. It can
utilize the operator's existing family plan database instead of
creating its own database.
[0064] 5. Buddy Connect: The Buddy Connect service allows a user to
create a buddy group and make an instant conference by any buddy
member in the buddy group.
[0065] 6. Quick Reach: The Quick Reach service is a call
originating service that allows a user to create a list of phone
numbers in order to reach a particular person. When the user
originates this type of call, all the phones for that particular
person are called and rang until one of the phones answers the
call, and then the rest of call attempts are dropped.
[0066] The Enhanced Group Calling Features include Voicemail
Diversion, Reverse Quick Reach, and Single Number Group
Calling:
[0067] A. Voicemail Diversion: The recognition of a diversion to
voicemail and dropping the leg.
[0068] B. Reverse Quick Reach: The terminating subscriber defines
how a call dialed to his/her mobile number should be handled and
directed. In addition, the RTX is removed from the media path
following a Quick Reach termination.
[0069] C. Single Number Group Calling: An optimal design
implementation of single number based group calling in a cellular
network including a geo-redundant RTX deployment.
[0070] 2 System Description
[0071] 2.1 Overview
[0072] The following illustration explains the network reference
architecture used to provide the Enhanced Group Calling Features
for the Connected Portfolio Services described herein. These
Enhanced Group Calling Features for the Connected Portfolio
Services are provided without any changes to the existing network
infrastructure, but merely the addition of a service control point,
known as a Real-Time Exchange (RTX), connected to the network and a
client application embedded in the handset (although a clientless
version of the handset may be provided as well).
[0073] 2.2 Network Architecture
[0074] FIG. 1 is a block diagram that illustrates an exemplary
embodiment of a wireless communications network according to a
preferred embodiment of the present invention.
[0075] Within the network 100, an RTX 102, also known as a Dispatch
Gateway (DG), communicates with a MSC (Mobile Switching Center) 104
and PSTN (Public Switched Telephone Network) 106 using
SS7--ISUP/WIN/CAMEL (Signaling System 7--Integrated Services
Digital Network User Part/Wireless Intelligent Network/Customized
Applications for Mobile Enhanced Logic) messages at a signaling
plane 108. A bearer path 110 implements a TDM (Time Division
Multiplexing) interface carrying PCM (Pulse Code Modulation) or TFO
(Tandem Free Operation) voice frames. Support for TFO in this path
110 is negotiated between a BSC (Base Station Controller) 112 and
the RTX 102 for each originating and terminating leg of an AVS
call. The use of TFO ensures high voice quality (as voice vocoder
conversion is avoided) between mobile-to-mobile calls.
[0076] When a subscriber originates an AVS call, the MSC 104 routes
the call to the RTX 102. The MSC 104 also requests the BSC 112 via
116 to establish a radio traffic path 118 with a mobile station
(MS) 120 (also known as a handset or mobile unit) via the BTS (Base
Transceiver Station) 122 (as it does for a normal cellular call).
At this time, the BSC 112 tries to negotiate TFO (if it is
supported) on a TDM link with the far end (in this case, the RTX
102).
[0077] At the same time (after the MSC 104 terminates the group
call request to the RTX 102), the RTX 102 identifies the
terminating group users and their numbers, which may comprise an
MS-ISDN (Mobile Station-Integrated Services Digital Network)
number, an IMSI (International Mobile Subscriber Identity) number,
or an MDN (Mobile Directory Number).
[0078] The RTX 102 sends an ISUP call origination request for each
terminating handset 120. It may send requests directly to the MSC
104, PSTN 106 or IP network 124 via a PDSN (Public Data Switched
Network) 126, Router 128, and/or Internet/Intranet 130, depending
on the routing table configuration for terminating numbers. Once
the bearer path 110 is established, the RTX 102 begins a
negotiation with the far end (in this case, the terminating BSC
112) for each terminating leg to a handset 120.
[0079] Once bearer paths 110 are established for originating and
terminating legs for an AVS call, the RTX 102 switches (or
duplicates) voice or data from the originating handset 120 to all
terminating handsets 120.
[0080] The RTX 102 may also use an IP network 124 or the
Internet/Intranet 130. The IP network 124 or the Internet/Intranet
130 can be used in a toll bypass mode where two RTXs 102 can
exchange voice traffic bypassing the PSTN 106. However, each RTX
102 is responsible for terminating traffic to its closest MSC 104.
In this case, the IP network 124 or the Internet/Intranet 130 is
used as a backbone transport of voice traffic between two RTXs
102.
[0081] The IP network 124 or the Internet/Intranet 130 can also be
used for a registration and presence application. Since the MSC 104
will not direct a registration request from a handset 120 to the
RTX 102 (because it would require changes in the MSC 104), the
latter does not have any information of the registered handset 120.
To circumvent this issue, a registration and presence application
runs over an IP stack in the handset 120. After the handset 120
registers for a data interface (i.e., obtaining an IP address) with
the PDSN 126 (or Serving GSM Service Nodes (SGSN) in the case of
GSM networks), the registration and presence application in the
handset 120 registers with the RTX 102 using its IP address. The
RTX 102 also uses this IP interface to update the presence
information of other group members to a handset 120.
[0082] An alternative embodiment may use the SMS (Short Message
Service) transport to carry presence messages over a data channel.
The RTX 102 interacts with the handset 120 using predefined
presence application related messages that are transported as SMS
messages. The same messages can be transported via the PDSN 126
interface, if group users have data service.
[0083] During roaming, a Home Location Register (HLR) 132 and
Visitor Location Register (VLR) 134 can be accessed via the MSC 104
and a MAP link 136. The HLR 132 and VLR 134 are used to track the
mobile handsets 120 within home or foreign networks, while the RTX
102 is used to track the presence of members of a group within the
home or foreign networks and updates the mobile handsets 120 for
those members with the network availability of other members of the
group.
[0084] A Short Message Service Center (SMSC) 138 is accessible via
the IP network 124 (or other element) for the storage of text
messages (SMS messages). When an SMS message is sent to a handset
120, the message is first stored in the SMSC 138 until the
recipient handset 120 is available (e.g., a store-and-forward
option).
[0085] 2.3 Real Time Exchange
[0086] FIG. 2 illustrates a proposed architecture for the RTX 102
according to the preferred embodiment of the present invention.
[0087] The architecture includes a Call Processing system 200,
Presence Server 202, Real-Time Event Processing system 204, one or
more Media Managers 206, and an SMPP (Short Message Peer-to-Peer)
Transport 208, as well as modules for various SS7 protocols, such
as MTP-1 (Message Transfer Part Level 1) 210, MTP-2 (Message
Transfer Part Level 2) 212, MTP-3 (Message Transfer Part Level 3)
214, ISUP (Integrated Services Digital Network User Part) 216, SCCP
(Signaling Connection Control Part) 218, and TCAP (Transactions
Capabilities Application Part) 220 protocols.
[0088] The Call Processing system 200, Presence Server 202, Media
Managers 204, SMPP Transport 206, and other modules communicate
across an IP network 222. The Real-Time Event Processing system 204
communicates directly with the Call Processing system 200, Presence
Server 202, and the modules for various SS7 protocols. The modules
for various SS7 protocols communicate with other entities via a SS7
Signaling Link 224. The SMPP Transport 206 communicates with a SMSC
(Short Message Service Center) gateway using the SMPP protocol 226.
The Media Managers 204 communicate among themselves using the H.110
protocol 228 (or some other protocol, such TCP/IP).
[0089] The operation of these various components are described in
more detail below, as well as in the co-pending and
commonly-assigned patent applications cross-referenced above and
incorporated by reference herein.
[0090] The originating handset 120 signals the RTX 102 via the
wireless network 100, e.g., by transmitting one or more messages to
the RTX 102. The Media Manager systems 206 receive the messages and
pass the messages to the Call Processing system 200. The Call
Processing (CP) system 200 determines whether the originating
handset 120 has subscribed to the AVS before originating the AVS
call. Upon confirmation, the Call Processing system 200 initiates a
new AVS call. The Call Processing system 200 interacts with the
Presence Server 202 and Real-Time Event Processing system 204 to
cause the wireless network 100 to perform call setup with the
terminating handsets 120 for the AVS call, and thereafter to manage
the AVS call.
[0091] During the AVS call, the Call Processing system 200
interacts with the Media Manager systems 206 to maintain the H.110
channels 227 and assign any additional H.110 channels 228 required
for the AVS call, which may span across multiple Media Manager
systems 206. During the AVS call, the Media Manager systems 206 of
the RTX 102 are used to mix audio streams between the originating
handset 120 and the terminating handset 120, and then deliver these
mixed audio streams to the originating handset 120 and the
terminating handset 120. The H.110 channels 228 are used for
passing mixed and unmixed audio streams voice between the Media
Manager systems 200 as required.
[0092] 2.4 Mobile Station Components
[0093] FIG. 3 illustrates the high-level functional components and
their interfaces in the handset 120 according to a preferred
embodiment of the present invention. In one embodiment, the
software architecture used in the handset 120 is based on an Open
OS implementation and is available under multiple operating
systems, including JAVA, WINDOWS MOBILE, SYMBIAN and BREW.
[0094] Preferably, the software architecture used in the handset
120 provides an application programming interface (API) that
supports the logic and data required within the handset 120 for
providing cellular service, including the functions necessary for
the making an AVS call generally, for providing the Connected
Portfolio Services generally and for providing the Enhanced Group
Calling Features for the Connected Portfolio Services
specifically.
[0095] The high-level functional components of the handset 120
include a subscriber identity module (SIM) 300, encoder/decoder
302, processing logic 304 and user interface 306. A client
application 308 is provided on the SIM 300 that supports AVS
functionality for the handset 120. In addition, the SIM 300 stores
a database 310, which includes an address book, AVS contacts and/or
group information.
[0096] At power-on, the handset 120 loads the client application
308 necessary to support the AVS. This functionality provided
includes the "look and feel" of the menu displays on the handset
120, as well as user interaction with the menu displays.
[0097] During operation, the encoder/decoder 302 decodes and
encodes messages, and populates specific data structures in the
handset 120. The encoder/decoder 302 checks the validity of the
incoming messages by verifying mandatory parameters for each of the
incoming messages. A message will not be processed further if the
encoder/decoder 302 fails to decode the message.
[0098] The processing logic 304 handles all the AVS related
functionalities. The processing logic 304 implementation is
device-specific and vendor-specific, and it interacts with the
other components, including the encoder/decoder 302, user interface
306, client application 308 and database 310.
[0099] The processing logic 304 provides an auto-answer mechanism
for AVS calls. Specifically, when a call is received, the
processing logic 304 automatically answers the call. The processing
logic 304 makes use of call notification for incoming call
detection and, based on various parameters received within the call
notification, determines whether the call is an AVS call. If the
call is an AVS call, then the processing logic 304 uses "AT"
commands to answer the AVS call and turn on the speaker of the
handset 120. (All of this takes place within a certain time
period.) On the other hand, if the call is not an AVS call, then
normal call processing is performed by the handset 120.
[0100] The processing logic 304 also provides "floor control" using
DTMF tone control. For example, in push-to-talk (P2T) calls, which
are half-duplex, a determination of who may talk is based on who
has the "floor." Using the processing logic 304 provided in the
handset 120, appropriate DTMF tones are sent to the RTX 102 in
accordance with specific key sequences (i.e., pressing and/or
releasing a P2T key) that indicate whether the "floor" has been
requested and/or released by the user.
[0101] In addition, the processing logic 304 provides SMS
destination control based on the type of subscriber. At the time of
subscriber data provisioning, if it is determined that the handset
120 will use AVS based logic, then appropriate logic is invoked in
the RTX 102 to send presence messages over SMS to the handset 120.
Similarly, the handset 120 is configured at the time of
provisioning to receive/accept such SMS and respond to the RTX 102
appropriately.
[0102] Finally, the processing logic 304 also enables subscribers
to track the presence of fellow members of the group in the network
100 on their handset 120, and provides a mechanism and API to
carry-out contacts and group management operations on the handset
120, such as add member, delete member, etc.
[0103] Since most of the presence information is stored in the
database 310, the database 310 is tightly integrated with the
processing logic 304. The database 310 stores groups, contacts,
presence and availability related information. The database 310
information essentially contains group and member information along
with presence information associated with each group and member.
Apart from group and member information, the database 310 also
stores subscriber information, such as privileges, presence
information, etc. The other components of the handset 120 may
interact with the database 310 to retrieve/update the group,
members and presence information for various operations. The
database 310 also has pointers to the native address book on the
handset 120, to provide seamless "alias" naming for contacts used
with cellular calls, as well as AVS.
[0104] The user interface 306 provides a mechanism for the user to
view and manage groups, group members, contacts, presence and
availability. The user interface 306 also makes it possible to
invoke the AVS from the group/contact list screens, as described in
more detail below.
[0105] 2.5 Connected Portfolio Services
[0106] The RTX 102 and handset 120 work together to provide the
functionality of the Connected Portfolio Services for the wireless
communications network 100. The specifics of this functionality are
described in more detail in the following sections.
[0107] 3 Scheduled Conference
[0108] The Scheduled Conference service allows a moderator to
schedule a conference in advance. Establishing a scheduled
conference can be done by connecting to the RTX 102 through the
handset 120 or via Internet access. The originator can specify how
to set up the type of participant connection (Dial-In or Dial-Out)
and whether a moderator (e.g., the originator) is required on the
call or not.
[0109] FIG. 4 illustrates the user interface 400 for the conference
scheduler as displayed on the handset 120, according to a preferred
embodiment of the present invention. The user can specify a subject
402, date 404, time 406, time zone 408, duration 410, as well as
dial mode options 412, including either Dial-In or Dial-Out modes
of operation.
[0110] The RTX 102 notifies each participant and originator with
the conference details using SMS. For Dial-Out conferences, also
known as an Instant Conference (IC), the RTX 102 dials each
participant at the scheduled time. For Dial-In conferences, also
known as a Reservationless Conference (RC), each participant simply
presses the Send key on their handset 120 after high-lighting the
bridge number within the conference details SMS.
[0111] FIG. 5 is a flowchart that illustrates the steps performed
in a Scheduled Conference, according to a preferred embodiment of
the present invention.
[0112] Block 500 represents the originator selecting a "New
Conference" option on the handset and providing the conference
details via the user interface shown in FIG. 4. The originator then
selects the conference participants from the address book, and
presses the Send key on the handset to complete the scheduling of
the conference, which sends an SMS to the RTX.
[0113] Block 502 represents the RTX sending a conference details
SMS to the conference participants.
[0114] Block 504 represents the initiation of the conference, at
the conference start time.
[0115] In Dial-In mode, the participant selects the conference
details SMS on the handset and then selects the Send key on the
handset to dial into the conference. The conference participants
may also dial in from a landline using a global number and access
code.
[0116] In Dial-Out mode, the RTX will dial out to all the
participants as well as the originator.
[0117] 3.1 End User Features
[0118] The main features of the Scheduled Conference include the
following: [0119] Dial-In or Dial-Out Conference Type, [0120] Start
Without Me option (Yes/No), [0121] Continue Without Me option
(Yes/No), [0122] Duration of Conference, and [0123] My Conferences
Tab (view of conferences originated and/or participated in).
[0124] 3.2 Mid Call Add/Drop
[0125] This feature provides the user with the ability to add or
drop participants to an active conference call. The user can select
some specified number of participants to add to or drop from a
conference call. The Mid Call Add/Drop feature can be accessed by
the user under an Options menu on the handset.
[0126] 3.3 Rejoin a Conference Call
[0127] This feature allows for the originator or any participants
of a conference call to rejoin an active conference call, if they
have dropped at any time. The RTX 102 sends an SMS to the
originator and all participants of the conference call with the
bridge information. The end user can simply press the Send key on
the handset 120 while displaying the SMS to rejoin the conference
call. To rejoin a clientless conference, the participants can dial
a global conference number and enter an access code at any time.
(An SMS is not sent to clientless conference participants.)
[0128] 3.4 Call Flows
[0129] The call flows for Scheduled Conference in CDMA and GSM
networks can be found in co-pending and commonly assigned U.S.
application Ser. No. 12/259,102, filed Oct. 27, 2008, by
Krishnakant M. Patel, Bruce Lawler, Gorachand Kundu, Ravi Ayyasamy,
Ravi Shankar Kumar, Harisha Mahabaleshwara Negalaguli, Basem Ahmad
Ardah, Prathap Chandana, Shan-Jen Chiou, Arun Velayudhan, and Ramu
Kandula, entitled "CONNECTED PORTFOLIO SERVICES FOR A WIRELESS
COMMUNICATIONS NETWORK," attorneys' docket number 154.32-US-U1,
which application claims the benefit under 35 U.S.C. Section 119(e)
of U.S. Provisional Application Ser. No. 60/982,650 (154.32-US-P1)
and U.S. Provisional Application Ser. No. 61/023,042
(154.32-US-P2), which applications are incorporated by reference
herein.
[0130] 4 Family Connect Group Call
[0131] The Family Connect service is an Instant Conference service
that utilizes the existing operator's existing family plan database
and terminates calls to all the family members when a nation-wide
number is dialed by the user.
[0132] The main features of Family Connect are: [0133] One group
per user, [0134] Existing family plan databases can be used, [0135]
One global nation-wide access number (i.e. a dialable number), and
[0136] Group management via the Internet.
[0137] Alternatively, when an operator's network does not provide a
family plan database, this service provides a Web interface for the
user to create/update/view her/his own family members.
[0138] 5 Buddy Connect Group Call
[0139] The Buddy Connect service is Instant Conference service,
wherein the user creates "buddy connect" groups via the
Internet.
[0140] The main features of Buddy Connect are: [0141] Multiple
groups per user, [0142] Each group contains up to a specified
number of members, [0143] Each member can be in multiple groups,
[0144] Each group is assigned a unique access number (i.e. a
dialable number), [0145] Group management is performed by the
creator via the Internet.
[0146] 6 Quick
[0147] Reach
[0148] The Quick Reach service allows a user to reach a called
party by making call attempts to all possible phone numbers (i.e.,
Customer Premise Equipment (CPE)) used or owned by the called
party. The user creates Quick Reach groups via the Internet.
[0149] The main features of Quick Reach are: [0150] Multiple groups
per user, [0151] Each group contains up to a specified number of
contact numbers, [0152] Each group is assigned a unique access
number (i.e. a dialable number), [0153] Group management is
performed by the creator via the Internet.
[0154] 7 Reservationless Conference/Clientless Conference
[0155] A Reservationless Conference, also known as Clientless
Conference, provides a "Meet me on my bridge" capability without a
client on the handset. Each clientless conference subscriber will
have a standing bridge that can be accessed at anytime. The
conference owner will create an access code for each conference and
provide the access code to the participants. Participants will
enter the access code that was created by the conference owner and
join the conference once the owner has joined the call.
[0156] 7.1 Originator User Flows
[0157] FIG. 6 is a flowchart that illustrates the steps performed
in a Reservationless Conference Origination, according to a
preferred embodiment of the present invention. Block 600 represents
the originator, using a handset with a client application, sending
an off-line conference notice, which includes a call-in number
allocated by the RTX, a conference ID such as the originator's
mobile number, and an access code.
[0158] Block 602 represents the initiation of the conference, at
the conference start time. The originator and the participants dial
the call-in number, and are prompted by the RTX to enter the
conference ID and the access code. The conference participants may
dial in from a landline as well as a handset. All of the
participants can rejoin the conference call at any time using the
same steps.
[0159] 7.2 End User Features
[0160] The main features of the Reservationless Conference include
the following: [0161] A single conference bridge number to
remember, [0162] Mid Call Add using dialed digits, [0163] Mid Call
Drop using dialed digits, [0164] A List of Participants sent via
SMS to all members in conference, [0165] Support for any Dial-able
Number, [0166] Rejoin a conference call, and [0167] Originator
creates access code per conference.
[0168] 7.3 Mid Call Add/Drop
[0169] This feature provides the user with the ability to add or
drop participants to/from an active Clientless Conference. The
originator can enter the full MDN of the participant to add or drop
from the keypad of the handset during an active Clientless
Conference
[0170] 8 Enhanced Group Calling Features
[0171] As noted above, the Connected Portfolio Services include
such services as Mobile Conferencing (i.e.,
Scheduled/Instant/Reservationless Conferencing), Family Connect,
Group Connect, and Quick Reach, all of which use a single number to
designate a logical group of numbers.
[0172] In the case of Mobile Conferencing, Family Connect and Group
Connect, the numbers represented are the conference participant
numbers, family or group members' numbers, respectively. In the
case of Quick Reach, the numbers represent one particular
subscriber's numbers (such as home, office, mobile, etc.)
[0173] As noted above, the Enhanced Group Calling features for
these Connected Portfolio Services include: [0174] the recognition
of a diversion to voicemail and dropping the leg; [0175] the
introduction of "Reverse Quick Reach" where the terminating
subscriber defines how a call dialed to his/her mobile number
should be handled and directed; [0176] the removal of the RTX 102
from the bearer path following a Quick Reach termination; and
[0177] an optimal design implementation of single number based
group calling in a cellular network including a geo-redundant RTX
102 deployment.
[0178] 8.1 Dropping a Voicemail Leg in Group Calling Scenarios
[0179] In all variants of the Connected Portfolio Services, it is
imperative that the RTX 102 initiating the terminating legs of the
Connected Portfolio Services recognizes that one or more of the
terminating legs of the Connected Portfolio Services has been
diverted to voicemail and drops the terminating legs that have been
diverted to voicemail from the Connected Portfolio Services. The
diversion of call legs to voicemail, and the dropping those call
legs from inclusion in a conference bridge, prevents the group
communications from being recorded on a voicemail, which entails
privacy issues, and also prevents the awkward interjection of
voicemail greetings and announcements into the group
communications.
[0180] The present invention's diversion to voicemail is based on
two methods: [0181] the receipt by the RTX 102 of a diversion
indicator and diversion number in a ISUP Call Progress Message
(CPG) message from the MSC 104, where the diversion number is then
compared and matched to a list of voicemail numbers by the RTX 102;
and [0182] the non-receipt of an indicator confirming participation
in the group call (e.g., requiring that the "#" key on the handset
120 be pressed by the terminating user upon answering a group
call).
[0183] FIG. 7 illustrates a first method for dropping a voicemail
leg in a group calling scenario, according to a preferred
embodiment of the present invention.
[0184] In Step 1, an originator (A) dials a number that represents
a logical group of numbers.
[0185] In Step 2, the RTX translates the dialed number to several
termination numbers to whom the call should be terminated (B, C and
D).
[0186] In Step 3, the termination device C is unreachable, so the
MSC diverts the call to voicemail and indicates the diversion to
the RTX in an ISUP CPG message containing a diversion or
"forwarded-to" number.
[0187] In Step 4, the RTX receives the "forwarded-to" number in the
ISUP CPG message and checks the number against a database of
voicemail destination numbers. If the diversion number is
configured in the voicemail destination number database, then this
leg of the call is dropped from inclusion in the service.
[0188] Note that, if the diversion number does not match a
voicemail destination number in the database, then the next check
performed by the RTX is the "#" input from user that applies, as
described in the next figure.
[0189] FIG. 8 illustrates a second method for dropping a voicemail
leg in a group calling scenario, according to a preferred
embodiment of the present invention.
[0190] In Step 1, an originator (A) dials a number that represents
a logical group of numbers.
[0191] In Step 2, the RTX translates the dialed number to several
termination numbers to whom the call should be terminated (B, C and
D).
[0192] In Step 3, the termination device D is unreachable, and a
landline switch diverts the call to voicemail, but no indication is
sent to the RTX indicate the call diversion.
[0193] In Step 4, the RTX waits for the user to press "#" to join
the call. Since this leg is connected to a voicemail, no such input
will be forthcoming. The RTX will timeout waiting for this input,
and drops this leg of the call from inclusion in the service.
[0194] 8.2 Reverse Quick Reach
[0195] As noted above, Quick Reach is a service that allows the
originator to ring multiple phone numbers of the same person
simultaneously, sequentially or selectively (i.e., an originating
user service). Reverse Quick Reach is variation of the above
service, albeit a terminating user service that applies termination
call handling to the terminator's "published mobile number" during
termination and allows simultaneous, sequential or selective
ringing of multiple phone numbers. Specifically, when the
terminating user answers one of the phone numbers, the remaining
legs of the call are dropped.
[0196] The service is accomplished by arming the mobile number for
terminating triggers. A Gateway MSC (GMSC) 104 terminating the
call, which is an MSC 104 directly connected to the RTX 102, is
instructed to connect to the RTX 102 for subscribers that are
configured for this service. The RTX 102 then terminates the calls
to all the numbers configured for this user. When the user answers
at one of the numbers (and presses the "#" key to confirm joining
the group call), the RTX 102 drops the remaining (unanswered) legs
and connects the user to the originator.
[0197] FIG. 9 illustrates a method for performing Reverse Quick
Reach, according to a preferred embodiment of the present
invention.
[0198] In Step 1, an originator (A) dials B's mobile number.
[0199] In Step 2, the GMSC queries the HLR for the terminating
subscriber B and receives the trigger information which instructs
the GMSC to contact the RTX.
[0200] In Step 3, the GMSC contacts the RTX, which checks B's
subscription to this service and instructs the GMSC to connect the
call to the RTX.
[0201] In Step 4, the RTX terminates the call to all of the
destination numbers configured for B (B, C and D). The answered leg
is preserved and the remaining unanswered legs are dropped from the
call.
[0202] Note that, if B is not configured on the RTX for any other
destination numbers, the RTX will leave itself out of the media
path by instructing the GMSC to "Continue" in its processing of the
call toward B (i.e., Step 3).
[0203] Note also that, termination to B from the RTX may trigger
another query to the RTX, just as in Step 3. For the second query,
the RTX instructs the GMSC to "Continue" processing the call toward
B.
[0204] 8.2.1 Quick Reach and Reverse Quick Reach Optimization
[0205] Both Quick Reach and Reverse Quick Reach are services that
ultimately connect an originator to one and only one terminator.
This presents an opportunity for bearer path optimization where the
RTX 102 can exclude itself from the bearer path after the call has
been answered (i.e., normal call processing in the network 100 can
be used).
[0206] To take advantage of this optimization, the terminating call
must traverse the MSC 104 that originated the call, which typically
is the GMSC 104 through which the RTX 102 is connected. The RTX 102
would instruct the GMSC 104 to initiate a connection "cut-through"
very similar to the cut-through that occurs during subsequent
handoff scenarios, where IS-41 message directives indicate to an
anchor MSC 104 to connect to a serving MSC 104 by dropping an
intervening MSC 104 from the connection; however, in this scenario,
it is the intervening RTX 102 that is dropped from the
connection.
[0207] FIG. 10 illustrates a method for performing Quick Reach and
Reverse Quick Reach optimization, according to a preferred
embodiment of the present invention.
[0208] In Step 1, an originator (A) dials a number for B that
represents a logical group of numbers.
[0209] In Step 2, the call is routed through the GMSC to the
RTX.
[0210] In Step 3, the RTX terminates the call to all of the
destination numbers configured for B (B and C). The answered leg
(B) is preserved and the remaining leg (C) is dropped from the
call.
[0211] In Step 4, knowing that B has answered the call, the RTX can
then issue a facility directive to the GMSC that the circuits that
need to be connected internally within the GMSC.
[0212] In Step 5, the GMSC reconfigures the connections to leave
the RTX out of the media path by connecting the incoming bearer
path or leg from A, represented by an associated incoming CIC
(Circuit Identification Code), to the outgoing bearer path or leg
to B, represented by an associated outgoing CIC.
[0213] Note that, if B is not configured on the RTX for more than
one destination numbers, the RTX will leave itself out of the media
path by instructing the GMSC to "Continue" in its processing of the
call toward B (Step 3).
[0214] Note also that, termination to B from the RTX may trigger
another query to RTX, just as in Step 3. For the second query, the
RTX instructs the GMSC to "Continue" processing call toward B.
[0215] 8.3 Single Number Based Group Calling
[0216] Each of the Enhanced Group Calling features, including
Mobile Conferencing, Family Connect, Group Connect and Quick Reach,
that employ a single number to represent a logical group of
numbers, can be realized in an network 100 using one of the
following approaches described herein.
[0217] These approaches include the deployment of a geo-redundant
RTX 102 (i.e., a Geo-RTX), which is a redundant RTX 102 located at
a geographically different location. Moreover, these approaches
revolve around an optimal use of a block of numbers and simplifying
Translation Table entries within the network 100 through a creative
use of a temporary routing number.
[0218] These approaches collectively apply to a family of features
known as "Connected Applications," and are defined by Connected
Applications Routing.
[0219] 8.3.1 Connected Applications Routing (Approach 1)
[0220] FIG. 11 illustrates a method for performing the single
number based group calling in a normal scenario when the RTX is
active, according to a preferred embodiment of the present
invention.
[0221] In Step 1, the user dials 111-222-3001 for Group 1.
[0222] In Step 2, this dialing string is analyzed by the MSC
(Originator: A, Dialed: 111-222-3001), and routed to RTX2.
[0223] In Step 3, RTX2 determines from an EMS (Element Management
System) Table that 111-222-3001 is assigned to RTX1. RTX2 queries
RTX1 to retrieve a temporary routing number.
[0224] In Step 4, RTX1 allocates a number from a temporary routing
number block and assigns it to the call from A.
[0225] In Step 5, RTX1 responds back to RTX2 with the allocated
number (444-222-6001).
[0226] In Step 6, the Analyzed Response is provided to the MSC
(Connect to: 444-222-6001).
[0227] In Step 7, using a Translation Table, the MSC sends an IAM
(Initial Address Message) message to RTX1 (Calling Party: A, Called
Party: 444-222-6001).
[0228] In Step 8, RTX1 obtains the original dialed digits ((A,
444-222-6001).fwdarw.111-222-3001), and then obtains the group
information based on (A, 111-222-3001).
[0229] In Step 9, the group call is completed or rejoined.
[0230] FIG. 12 illustrates a method for performing the single
number based group calling in a geo-redundancy approach when the
RTX is unavailable, according to a preferred embodiment of the
present invention.
[0231] In Step 1, the user dials 111-222-3001 for Group 1.
[0232] In Step 2, this dialing string is analyzed by the MSC
(Originator: A, Dialed: 111-222-3001), and routed to RTX2.
[0233] In Step 3, RTX2 determines from the EMS Table that
111-222-3001 is assigned to RTX1, but RTX1 is down (unavailable).
RTX2 queries Geo-RTX2 to retrieve a temporary routing numbers.
[0234] In Step 4, Geo-RTX2 allocates a number from the temporary
routing number block and assigns it to the call from A.
[0235] In Step 5, Geo-RTX2 responds back to RTX2 with the allocated
number (666-777-1007).
[0236] In Step 6, the Analyzed Response is provided to the MSC
(Connect to: 666-777-1007).
[0237] In Step 7, using the Translation Table, the MSC sends an IAM
message to Geo-RTX2 (Calling Party: A, Called Party:
666-777-1007).
[0238] In Step 8, Geo-RTX2 obtains the group information based on
((A, 666-777-1007).fwdarw.111-222-3001).
[0239] In Step 9, the group call is completed or rejoined.
[0240] FIG. 13 illustrates a method for performing the single
number based group calling in a geo-redundancy approach when the
RTX is unavailable and a "race condition" exists due to a second
call from the same user, according to a preferred embodiment of the
present invention.
[0241] In Step 1, the user dials 111-222-3002 for Group 2.
[0242] In Step 2, this dialing string is analyzed by the MSC
(Originator: A, Dialed: 111-222-3002), and routed to RTX2.
[0243] In Step 3, RTX2 determines from the EMS Table that
111-222-3002 is assigned to RTX1, but RTX1 is down (unavailable),
so RTX2 forwards the call to Geo-RTX2, based on the EMS Table.
[0244] In Step 4, Geo-RTX2 allocates a number from the temporary
routing number block and assigns it to the call from A.
[0245] In Step 5, Geo-RTX2 responds back to RTX2 with the allocated
number (666-777-1008).
[0246] In Step 6, the Analyzed Response is provided to the MSC
(Connect to: 666-777-1008).
[0247] In Step 7, using the Translation Table, the MSC sends an IAM
message to Geo-RTX2 (Calling Party: A, Called Party:
666-777-1008).
[0248] In Step 8, Geo-RTX2 obtains the group information based on
((A, 666-777-1008).fwdarw.111-222-3002).
[0249] In Step 9, the group call is completed or rejoined.
[0250] 8.3.2 Connected Applications Routing (Approach 2)
[0251] FIG. 15 illustrates a method for performing the single
number based group calling in a normal scenario when the RTX is
active, according to a preferred embodiment of the present
invention.
[0252] In Step 1, the user dials 111-222-3001 for Group 1.
[0253] In Step 2, this dialing string is analyzed by the MSC
(Originator: A, Dialed: 111-222-3001), and routed to RTX1.
[0254] In Step 3, RTX1 verifies the subscription for A and
generates an Analyzed Response including a routing number that
connects to itself, based on the EMS table.
[0255] In Step 4, RTX1 sends the Analyzed Response to the MSC
(Connect to: 111-222-3001).
[0256] In Step 5, using the Translation Table, the MSC sends an IAM
message to RTX1 (Calling Party: A, Called Party: 111-222-3001).
[0257] In Step 6, RTX1 obtains the group information based on (A,
111-222-3001).
[0258] In Step 7, the group call is completed or rejoined.
[0259] FIG. 16 illustrates a method for performing the single
number based group calling in a geo-redundancy approach when the
RTX is unavailable.
[0260] In Step 1, the user dials 111-222-3001 for Group 1.
[0261] In Step 2, this dialing string is analyzed by the MSC
(Originator: A, Dialed: 111-222-3001). The MSC determines that RTX1
is unavailable, and forwards the trigger to Geo-RTX1.
[0262] In Step 3, Geo-RTX1 verifies the subscription for A and
generates an Analyzed Response including a routing number from the
temporary routing number block that connects to itself, based on
the EMS, and assigns the routing number to the call from A.
[0263] In Step 4, Geo-RTX1 sends the Analyzed Response to the MSC
(Connect to: 666-777-1001).
[0264] In Step 5, using the Translation Table, the MSC sends an IAM
message to Geo-RTX1 (Calling Party: A, Called Party:
666-777-1001).
[0265] In Step 6, Geo-RTX1 obtains the group information based on
(666-777-1001.fwdarw.(A, 111-222-3001).
[0266] In Step 7, the group call is completed or rejoined.
[0267] 8.3.3 Connected Applications Routing (Approach 3)
[0268] FIG. 17 illustrates a method for performing the single
number based group calling in a geo-redundancy approach when the
RTX is active, according to a preferred embodiment of the present
invention.
[0269] In Step 1, the user dials 111-222-3001 for Group 1.
[0270] In Step 2, using the Translation Table, the MSC sends an IAM
message to RTX1 (Calling Party: A, Called Party: 111-222-3001).
[0271] In Step 3, RTX1 obtains the group information based on (A,
111-222-3001).
[0272] In Step 4, the group call is completed or rejoined.
[0273] FIG. 18 illustrates a method for performing the single
number based group calling in a geo-redundancy approach when the
RTX is unavailable, according to a preferred embodiment of the
present invention.
[0274] In Step 1, the user dials 111-222-3001 for Group 1.
[0275] In Step 2, this dialing string is analyzed by the MSC
(Originator: A, Dialed: 111-222-3001), using the Translation Table,
which identifies RTX1. However, the MSC determines that RTX1 is
unavailable. Using the Call Forwarding Table, the MSC sends an IAM
message to Geo-RTX1 (Calling Party: A, Called Party: 666-777-3001,
Original Dialed Number: 111-222-3001).
[0276] In Step 3, Geo-RTX1 obtains the group information based on
(A, 111-222-3001).
[0277] In Step 4, the group call is completed or rejoined.
[0278] 15 Conclusion
[0279] The foregoing description of the preferred embodiment of the
invention has been presented for the purposes of illustration and
description. It is not intended to be exhaustive or to limit the
invention to the precise form disclosed. Many modifications and
variations are possible in light of the above teaching. It is
intended that the scope of the invention be limited not with this
detailed description, but rather by the claims appended hereto.
* * * * *